RTP Pro le for Audio and Video Conferences with Minimal Control

Network Working Group H. SchulzrinneRequest for Comments: 3551 Columbia UniversityObsoletes: 1890 S. CasnerCategory: Standards Track Packet Design

July 2003

RTP Profile for Audio and Video Conferences with Minimal Control

Status of this Memo

This document specifies an Internet standards track protocol for the Internet community, andrequests discussion and suggestions for improvements. Please refer to the current edition of the“Internet Official Protocol Standards” (STD 1) for the standardization state and status of thisprotocol. Distribution of this memo is unlimited.

Copyright Notice

Copyright (C) The Internet Society (2003). All Rights Reserved.

Abstract

This document describes a profile called “RTP/AVP” for the use of the real-time transport protocol(RTP), version 2, and the associated control protocol, RTCP, within audio and video multipartici-pant conferences with minimal control. It provides interpretations of generic fields within the RTPspecification suitable for audio and video conferences. In particular, this document defines a set ofdefault mappings from payload type numbers to encodings.

This document also describes how audio and video data may be carried within RTP. It defines a setof standard encodings and their names when used within RTP. The descriptions provide pointersto reference implementations and the detailed standards. This document is meant as an aid forimplementors of audio, video and other real-time multimedia applications.

This memorandum obsoletes RFC 1890. It is mostly backwards-compatible except for functionsremoved because two interoperable implementations were not found. The additions to RFC 1890codify existing practice in the use of payload formats under this profile and include new payloadformats defined since RFC 1890 was published.

Schulzrinne & Casner Standards Track []

RFC 3551 RTP A/V Profile July 2003

Table of Contents

1. Introduction 4

1.1 Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

2. RTP and RTCP Packet Forms and Protocol Behavior 4

3. Registering Additional Encodings 6

4. Audio 7

4.1 Encoding-Independent Rules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

4.2 Operating Recommendations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

4.3 Guidelines for Sample-Based Audio Encodings . . . . . . . . . . . . . . . . . . . . . 9

4.4 Guidelines for Frame-Based Audio Encodings . . . . . . . . . . . . . . . . . . . . . 9

4.5 Audio Encodings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10

4.5.1 DVI4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

4.5.2 G722 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

4.5.3 G723 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

4.5.4 G726-40, G726-32, G726-24, and G726-16 . . . . . . . . . . . . . . . . . . . 15

4.5.5 G728 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16

4.5.6 G729 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16

4.5.7 G729D and G729E . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18

4.5.8 GSM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

4.5.9 GSM-EFR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

4.5.10 L8 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

4.5.11 L16 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

4.5.12 LPC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

4.5.13 MPA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

4.5.14 PCMA and PCMU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

4.5.15 QCELP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

4.5.16 RED . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

4.5.17 VDVI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24



5. Video 25

5.1 CelB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

5.2 JPEG . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

5.3 H261 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

5.4 H263 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

5.5 H263-1998 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

5.6 MPV . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

5.7 MP2T . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

5.8 nv . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

6. Payload Type Definitions 27

7. RTP over TCP and Similar Byte Stream Protocols 29

8. Port Assignment 29

9. Changes from RFC 1890 30

10. Security Considerations 32

11. IANA Considerations 33

12. References 33

12.1 Normative References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

12.2 Informative References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

13. Current Locations of Related Resources 34

14. Acknowledgments 36

15. Intellectual Property Rights Statement 36

16. Authors’ Addresses 37

17. Full Copyright Statement 38



1. Introduction

This profile defines aspects of RTP left unspecified in the RTP Version 2 protocol definition(RFC 3550) [1]. This profile is intended for the use within audio and video conferences withminimal session control. In particular, no support for the negotiation of parameters or member-ship control is provided. The profile is expected to be useful in sessions where no negotiation ormembership control are used (e.g., using the static payload types and the membership indicationsprovided by RTCP), but this profile may also be useful in conjunction with a higher-level controlprotocol.

Use of this profile may be implicit in the use of the appropriate applications; there may be noexplicit indication by port number, protocol identifier or the like. Applications such as sessiondirectories may use the name for this profile specified in Section 11.

Other profiles may make different choices for the items specified here.

This document also defines a set of encodings and payload formats for audio and video. Thesepayload format descriptions are included here only as a matter of convenience since they are toosmall to warrant separate documents. Use of these payload formats is NOT REQUIRED to usethis profile. Only the binding of some of the payload formats to static payload type numbers inTables 4 and 5 is normative.

1.1 Terminology

The key words “must”, “must not”, “required”, “shall”, “shall not”, “should”, “shouldnot”, “recommended”, “may”, and “optional” in this document are to be interpreted as de-scribed in RFC 2119 [2] and indicate requirement levels for implementations compliant with thisRTP profile.

This document defines the term media type as dividing encodings of audio and video content intothree classes: audio, video and audio/video (interleaved).

2. RTP and RTCP Packet Forms and Protocol Behavior

The section “RTP Profiles and Payload Format Specifications” of RFC 3550 enumerates a numberof items that can be specified or modified in a profile. This section addresses these items. Generally,this profile follows the default and/or recommended aspects of the RTP specification.

RTP data header: The standard format of the fixed RTP data header is used (one marker bit).

Payload types: Static payload types are defined in Section 6.

RTP data header additions: No additional fixed fields are appended to the RTP data header.

RTP data header extensions: No RTP header extensions are defined, but applications operat-ing under this profile may use such extensions. Thus, applications should not assume thatthe RTP header X bit is always zero and should be prepared to ignore the header extension.



If a header extension is defined in the future, that definition must specify the contents of thefirst 16 bits in such a way that multiple different extensions can be identified.

RTCP packet types: No additional RTCP packet types are defined by this profile specification.

RTCP report interval: The suggested constants are to be used for the RTCP report intervalcalculation. Sessions operating under this profile may specify a separate parameter for theRTCP traffic bandwidth rather than using the default fraction of the session bandwidth.The RTCP traffic bandwidth may be divided into two separate session parameters for thoseparticipants which are active data senders and those which are not. Following the recom-mendation in the RTP specification [1] that 1/4 of the RTCP bandwidth be dedicated todata senders, the recommended default values for these two parameters would be 1.25%and 3.75%, respectively. For a particular session, the RTCP bandwidth for non-data-sendersmay be set to zero when operating on unidirectional links or for sessions that don’t requirefeedback on the quality of reception. The RTCP bandwidth for data senders should bekept non-zero so that sender reports can still be sent for inter-media synchronization and toidentify the source by CNAME. The means by which the one or two session parameters forRTCP bandwidth are specified is beyond the scope of this memo.

SR/RR extension: No extension section is defined for the RTCP SR or RR packet.

SDES use: Applications may use any of the SDES items described in the RTP specification.While CNAME information must be sent every reporting interval, other items should onlybe sent every third reporting interval, with NAME sent seven out of eight times within thatslot and the remaining SDES items cyclically taking up the eighth slot, as defined in Section6.2.2 of the RTP specification. In other words, NAME is sent in RTCP packets 1, 4, 7, 10,13, 16, 19, while, say, EMAIL is used in RTCP packet 22.

Security: The RTP default security services are also the default under this profile.

String-to-key mapping: No mapping is specified by this profile.

Congestion: RTP and this profile may be used in the context of enhanced network service, forexample, through Integrated Services (RFC 1633) [4] or Differentiated Services (RFC 2475)[5], or they may be used with best effort service.

If enhanced service is being used, RTP receivers should monitor packet loss to ensure thatthe service that was requested is actually being delivered. If it is not, then they shouldassume that they are receiving best-effort service and behave accordingly.

If best-effort service is being used, RTP receivers should monitor packet loss to ensure thatthe packet loss rate is within acceptable parameters. Packet loss is considered acceptableif a TCP flow across the same network path and experiencing the same network conditionswould achieve an average throughput, measured on a reasonable timescale, that is not lessthan the RTP flow is achieving. This condition can be satisfied by implementing congestioncontrol mechanisms to adapt the transmission rate (or the number of layers subscribed for alayered multicast session), or by arranging for a receiver to leave the session if the loss rateis unacceptably high.



The comparison to TCP cannot be specified exactly, but is intended as an “order-of-magnitude”comparison in timescale and throughput. The timescale on which TCP throughput is mea-sured is the round-trip time of the connection. In essence, this requirement states that it isnot acceptable to deploy an application (using RTP or any other transport protocol) on thebest-effort Internet which consumes bandwidth arbitrarily and does not compete fairly withTCP within an order of magnitude.

Underlying protocol: The profile specifies the use of RTP over unicast and multicast UDP aswell as TCP. (This does not preclude the use of these definitions when RTP is carried byother lower-layer protocols.)

Transport mapping: The standard mapping of RTP and RTCP to transport-level addresses isused.

Encapsulation: This profile leaves to applications the specification of RTP encapsulation in pro-tocols other than UDP.

3. Registering Additional Encodings

This profile lists a set of encodings, each of which is comprised of a particular media data compres-sion or representation plus a payload format for encapsulation within RTP. Some of those payloadformats are specified here, while others are specified in separate RFCs. It is expected that addi-tional encodings beyond the set listed here will be created in the future and specified in additionalpayload format RFCs.

This profile also assigns to each encoding a short name which may be used by higher-level controlprotocols, such as the Session Description Protocol (SDP), RFC 2327 [6], to identify encodingsselected for a particular RTP session.

In some contexts it may be useful to refer to these encodings in the form of a MIME content-type.To facilitate this, RFC 3555 [7] provides registrations for all of the encodings names listed here asMIME subtype names under the “audio” and “video” MIME types through the MIME registrationprocedure as specified in RFC 2048 [8].

Any additional encodings specified for use under this profile (or others) may also be assigned namesregistered as MIME subtypes with the Internet Assigned Numbers Authority (IANA). This registryprovides a means to insure that the names assigned to the additional encodings are kept unique.RFC 3555 specifies the information that is required for the registration of RTP encodings.

In addition to assigning names to encodings, this profile also assigns static RTP payload typenumbers to some of them. However, the payload type number space is relatively small and cannotaccommodate assignments for all existing and future encodings. During the early stages of RTPdevelopment, it was necessary to use statically assigned payload types because no other mechanismhad been specified to bind encodings to payload types. It was anticipated that non-RTP meansbeyond the scope of this memo (such as directory services or invitation protocols) would be specifiedto establish a dynamic mapping between a payload type and an encoding. Now, mechanisms fordefining dynamic payload type bindings have been specified in the Session Description Protocol(SDP) and in other protocols such as ITU-T Recommendation H.323/H.245. These mechanisms



associate the registered name of the encoding/payload format, along with any additional requiredparameters, such as the RTP timestamp clock rate and number of channels, with a payload typenumber. This association is effective only for the duration of the RTP session in which the dynamicpayload type binding is made. This association applies only to the RTP session for which it is made,thus the numbers can be re-used for different encodings in different sessions so the number spacelimitation is avoided.

This profile reserves payload type numbers in the range 96-127 exclusively for dynamic assignment.Applications should first use values in this range for dynamic payload types. Those applicationswhich need to define more than 32 dynamic payload types may bind codes below 96, in which caseit is recommended that unassigned payload type numbers be used first. However, the staticallyassigned payload types are default bindings and may be dynamically bound to new encodings ifneeded. Redefining payload types below 96 may cause incorrect operation if an attempt is made tojoin a session without obtaining session description information that defines the dynamic payloadtypes.

Dynamic payload types should not be used without a well-defined mechanism to indicate themapping. Systems that expect to interoperate with others operating under this profile shouldnot make their own assignments of proprietary encodings to particular, fixed payload types.

This specification establishes the policy that no additional static payload types will be assignedbeyond the ones defined in this document. Establishing this policy avoids the problem of tryingto create a set of criteria for accepting static assignments and encourages the implementation anddeployment of the dynamic payload type mechanisms.

The final set of static payload type assignments is provided in Tables 4 and 5.

4. Audio

4.1 Encoding-Independent Rules

Since the ability to suppress silence is one of the primary motivations for using packets to transmitvoice, the RTP header carries both a sequence number and a timestamp to allow a receiver todistinguish between lost packets and periods of time when no data was transmitted. Discontiguoustransmission (silence suppression) may be used with any audio payload format. Receivers mustassume that senders may suppress silence unless this is restricted by signaling specified elsewhere.(Even if the transmitter does not suppress silence, the receiver should be prepared to handle periodswhen no data is present since packets may be lost.)

Some payload formats (see Sections 4.5.3 and 4.5.6) define a “silence insertion descriptor” or “com-fort noise” frame to specify parameters for artificial noise that may be generated during a periodof silence to approximate the background noise at the source. For other payload formats, a genericComfort Noise (CN) payload format is specified in RFC 3389 [9]. When the CN payload formatis used with another payload format, different values in the RTP payload type field distinguishcomfort-noise packets from those of the selected payload format.

For applications which send either no packets or occasional comfort-noise packets during silence,the first packet of a talkspurt, that is, the first packet after a silence period during which packets



have not been transmitted contiguously, should be distinguished by setting the marker bit in theRTP data header to one. The marker bit in all other packets is zero. The beginning of a talkspurtmay be used to adjust the playout delay to reflect changing network delays. Applications withoutsilence suppression must set the marker bit to zero.

The RTP clock rate used for generating the RTP timestamp is independent of the number ofchannels and the encoding; it usually equals the number of sampling periods per second. For N -channel encodings, each sampling period (say, 1/8,000 of a second) generates N samples. (Thisterminology is standard, but somewhat confusing, as the total number of samples generated persecond is then the sampling rate times the channel count.)

If multiple audio channels are used, channels are numbered left-to-right, starting at one. In RTPaudio packets, information from lower-numbered channels precedes that from higher-numberedchannels. For more than two channels, the convention followed by the AIFF-C audio interchangeformat should be followed [3], using the following notation, unless some other convention is spec-ified for a particular encoding or payload format:

l leftr rightc centerS surroundF frontR rear

channels description channel1 2 3 4 5 6

2 stereo l r3 l r c4 l c r S5 Fl Fr Fc Sl Sr6 l lc c r rc S

Note: RFC 1890 defined two conventions for the ordering of four audio channels. Sincethe ordering is indicated implicitly by the number of channels, this was ambiguous. Inthis revision, the order described as “quadrophonic” has been eliminated to remove theambiguity. This choice was based on the observation that quadrophonic consumer audioformat did not become popular whereas surround-sound subsequently has.

Samples for all channels belonging to a single sampling instant must be within the same packet.The interleaving of samples from different channels depends on the encoding. General guidelinesare given in Section 4.3 and 4.4.

The sampling frequency should be drawn from the set: 8,000, 11,025, 16,000, 22,050, 24,000,32,000, 44,100 and 48,000 Hz. (Older Apple Macintosh computers had a native sample rate of22,254.54 Hz, which can be converted to 22,050 with acceptable quality by dropping 4 samples ina 20 ms frame.) However, most audio encodings are defined for a more restricted set of samplingfrequencies. Receivers should be prepared to accept multi-channel audio, but may choose to onlyplay a single channel.



4.2 Operating Recommendations

The following recommendations are default operating parameters. Applications should be pre-pared to handle other values. The ranges given are meant to give guidance to application writers,allowing a set of applications conforming to these guidelines to interoperate without additionalnegotiation. These guidelines are not intended to restrict operating parameters for applicationsthat can negotiate a set of interoperable parameters, e.g., through a conference control protocol.

For packetized audio, the default packetization interval should have a duration of 20 ms or oneframe, whichever is longer, unless otherwise noted in Table 1 (column “ms/packet”). The pack-etization interval determines the minimum end-to-end delay; longer packets introduce less headeroverhead but higher delay and make packet loss more noticeable. For non-interactive applicationssuch as lectures or for links with severe bandwidth constraints, a higher packetization delay maybe used. A receiver should accept packets representing between 0 and 200 ms of audio data. (Forframed audio encodings, a receiver should accept packets with a number of frames equal to 200ms divided by the frame duration, rounded up.) This restriction allows reasonable buffer sizing forthe receiver.

4.3 Guidelines for Sample-Based Audio Encodings

In sample-based encodings, each audio sample is represented by a fixed number of bits. Within thecompressed audio data, codes for individual samples may span octet boundaries. An RTP audiopacket may contain any number of audio samples, subject to the constraint that the number ofbits per sample times the number of samples per packet yields an integral octet count. Fractionalencodings produce less than one octet per sample.

The duration of an audio packet is determined by the number of samples in the packet.

For sample-based encodings producing one or more octets per sample, samples from different chan-nels sampled at the same sampling instant should be packed in consecutive octets. For example,for a two-channel encoding, the octet sequence is (left channel, first sample), (right channel, firstsample), (left channel, second sample), (right channel, second sample), . . . . For multi-octet encod-ings, octets should be transmitted in network byte order (i.e., most significant octet first).

The packing of sample-based encodings producing less than one octet per sample is encoding-specific.

The RTP timestamp reflects the instant at which the first sample in the packet was sampled, thatis, the oldest information in the packet.

4.4 Guidelines for Frame-Based Audio Encodings

Frame-based encodings encode a fixed-length block of audio into another block of compressed data,typically also of fixed length. For frame-based encodings, the sender may choose to combine severalsuch frames into a single RTP packet. The receiver can tell the number of frames contained in anRTP packet, if all the frames have the same length, by dividing the RTP payload length by theaudio frame size which is defined as part of the encoding. This does not work when carrying framesof different sizes unless the frame sizes are relatively prime. If not, the frames must indicate theirsize.



For frame-based codecs, the channel order is defined for the whole block. That is, for two-channelaudio, right and left samples should be coded independently, with the encoded frame for the leftchannel preceding that for the right channel.

All frame-oriented audio codecs should be able to encode and decode several consecutive frameswithin a single packet. Since the frame size for the frame-oriented codecs is given, there is noneed to use a separate designation for the same encoding, but with different number of frames perpacket.

RTP packets shall contain a whole number of frames, with frames inserted according to age withina packet, so that the oldest frame (to be played first) occurs immediately after the RTP packetheader. The RTP timestamp reflects the instant at which the first sample in the first frame wassampled, that is, the oldest information in the packet.

4.5 Audio Encodings

name of sampling defaultencoding sample/frame bits/sample rate ms/frame ms/packetDVI4 sample 4 var. 20G722 sample 8 16,000 20G723 frame N/A 8,000 30 30G726-40 sample 5 8,000 20G726-32 sample 4 8,000 20G726-24 sample 3 8,000 20G726-16 sample 2 8,000 20G728 frame N/A 8,000 2.5 20G729 frame N/A 8,000 10 20G729D frame N/A 8,000 10 20G729E frame N/A 8,000 10 20GSM frame N/A 8,000 20 20GSM-EFR frame N/A 8,000 20 20L8 sample 8 var. 20L16 sample 16 var. 20LPC frame N/A 8,000 20 20MPA frame N/A var. var.PCMA sample 8 var. 20PCMU sample 8 var. 20QCELP frame N/A 8,000 20 20VDVI sample var. var. 20

Table 1: Properties of Audio Encodings (N/A: not applicable; var.: variable)

The characteristics of the audio encodings described in this document are shown in Table 1; theyare listed in order of their payload type in Table 4. While most audio codecs are only specifiedfor a fixed sampling rate, some sample-based algorithms (indicated by an entry of “var.” in the



sampling rate column of Table 1) may be used with different sampling rates, resulting in differentcoded bit rates. When used with a sampling rate other than that for which a static payload type isdefined, non-RTP means beyond the scope of this memo must be used to define a dynamic payloadtype and must indicate the selected RTP timestamp clock rate, which is usually the same as thesampling rate for audio.

4.5.1 DVI4

DVI4 uses an adaptive delta pulse code modulation (ADPCM) encoding scheme that was specifiedby the Interactive Multimedia Association (IMA) as the “IMA ADPCM wave type”. However, theencoding defined here as DVI4 differs in three respects from the IMA specification:

• The RTP DVI4 header contains the predicted value rather than the first sample value con-tained the IMA ADPCM block header.

• IMA ADPCM blocks contain an odd number of samples, since the first sample of a blockis contained just in the header (uncompressed), followed by an even number of compressedsamples. DVI4 has an even number of compressed samples only, using the ‘predict’ word fromthe header to decode the first sample.

• For DVI4, the 4-bit samples are packed with the first sample in the four most significantbits and the second sample in the four least significant bits. In the IMA ADPCM codec, thesamples are packed in the opposite order.

Each packet contains a single DVI block. This profile only defines the 4-bit-per-sample version,while IMA also specified a 3-bit-per-sample encoding.

The “header” word for each channel has the following structure:

int16 predict; /* predicted value of first samplefrom the previous block (L16 format) */

u_int8 index; /* current index into stepsize table */u_int8 reserved; /* set to zero by sender, ignored by receiver */

Each octet following the header contains two 4-bit samples, thus the number of samples per packetmust be even because there is no means to indicate a partially filled last octet.

Packing of samples for multiple channels is for further study.

The IMA ADPCM algorithm was described in the document IMA Recommended Practices forEnhancing Digital Audio Compatibility in Multimedia Systems (version 3.0). However, the Inter-active Multimedia Association ceased operations in 1997. Resources for an archived copy of thatdocument and a software implementation of the RTP DVI4 encoding are listed in Section 13.



4.5.2 G722

G722 is specified in ITU-T Recommendation G.722, “7 kHz audio-coding within 64 kbit/s”. TheG.722 encoder produces a stream of octets, each of which shall be octet-aligned in an RTP packet.The first bit transmitted in the G.722 octet, which is the most significant bit of the higher sub-bandsample, shall correspond to the most significant bit of the octet in the RTP packet.

Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for theG722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 andmust remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000Hz.

4.5.3 G723

G723 is specified in ITU Recommendation G.723.1, “Dual-rate speech coder for multimedia com-munications transmitting at 5.3 and 6.3 kbit/s”. The G.723.1 5.3/6.3 kbit/s codec was defined bythe ITU-T as a mandatory codec for ITU-T H.324 GSTN videophone terminal applications. Thealgorithm has a floating point specification in Annex B to G.723.1, a silence compression algorithmin Annex A to G.723.1 and a scalable channel coding scheme for wireless applications in G.723.1Annex C.

This Recommendation specifies a coded representation that can be used for compressing the speechsignal component of multi-media services at a very low bit rate. Audio is encoded in 30 ms frames,with an additional delay of 7.5 ms due to look-ahead. A G.723.1 frame can be one of three sizes:24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s frame), or 4 octets. These 4-octet frames are calledSID frames (Silence Insertion Descriptor) and are used to specify comfort noise parameters. Thereis no restriction on how 4, 20, and 24 octet frames are intermixed. The least significant two bits ofthe first octet in the frame determine the frame size and codec type:

bits content octets/frame00 high-rate speech (6.3 kb/s) 2401 low-rate speech (5.3 kb/s) 2010 SID frame 411 reserved

It is possible to switch between the two rates at any 30 ms frame boundary. Both (5.3 kb/s and 6.3kb/s) rates are a mandatory part of the encoder and decoder. Receivers must accept both datarates and must accept SID frames unless restriction of these capabilities has been signaled. TheMIME registration for G723 in RFC 3555 [7] specifies parameters that may be used with MIME orSDP to restrict to a single data rate or to restrict the use of SID frames. This coder was optimizedto represent speech with near-toll quality at the above rates using a limited amount of complexity.

The packing of the encoded bit stream into octets and the transmission order of the octets isspecified in Rec. G.723.1 and is the same as that produced by the G.723 C code reference im-plementation. For the 6.3 kb/s data rate, this packing is illustrated as follows, where the header(HDR) bits are always “0 0” as shown in Fig. 1 to indicate operation at 6.3 kb/s, and the Z bitis always set to zero. The diagrams show the bit packing in “network byte order”, also known as



big-endian order. The bits of each 32-bit word are numbered 0 to 31, with the most significantbit on the left and numbered 0. The octets (bytes) of each word are transmitted most significantoctet first. The bits of each data field are numbered in the order of the bit stream representationof the encoding (least significant bit first). The vertical bars indicate the boundaries between fieldfragments.

0 1 2 30 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| LPC |HDR| LPC | LPC | ACL0 |LPC|| | | | | | ||0 0 0 0 0 0|0 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2||5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| ACL2 |ACL|A| GAIN0 |ACL|ACL| GAIN0 | GAIN1 || | 1 |C| | 3 | 2 | | ||0 0 0 0 0|0 0|0|0 0 0 0|0 0|0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0||4 3 2 1 0|1 0|6|3 2 1 0|1 0|6 5|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| GAIN2 | GAIN1 | GAIN2 | GAIN3 | GRID | GAIN3 || | | | | | ||0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0||3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|3 2 1 0|1 0 9 8|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| MSBPOS |Z|POS| MSBPOS | POS0 |POS| POS0 || | | 0 | | | 1 | ||0 0 0 0 0 0 0|0|0 0|1 1 1 0 0 0|0 0 0 0 0 0 0 0|0 0|1 1 1 1 1 1||6 5 4 3 2 1 0| |1 0|2 1 0 9 8 7|9 8 7 6 5 4 3 2|1 0|5 4 3 2 1 0|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| POS1 | POS2 | POS1 | POS2 | POS3 | POS2 || | | | | | ||0 0 0 0 0 0 0 0|0 0 0 0|1 1 1 1|1 1 0 0 0 0 0 0|0 0 0 0|1 1 1 1||9 8 7 6 5 4 3 2|3 2 1 0|3 2 1 0|1 0 9 8 7 6 5 4|3 2 1 0|5 4 3 2|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| POS3 | PSIG0 |POS|PSIG2| PSIG1 | PSIG3 |PSIG2|| | | 3 | | | | ||1 1 0 0 0 0 0 0|0 0 0 0 0 0|1 1|0 0 0|0 0 0 0 0|0 0 0 0 0|0 0 0||1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|2 1 0|4 3 2 1 0|4 3 2 1 0|5 4 3|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Figure 1: G.723 (6.3 kb/s) bit packing

For the 5.3 kb/s data rate, the header (HDR) bits are always “0 1”, as shown in Fig. 2, to indicateoperation at 5.3 kb/s.



0 1 2 30 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| LPC |HDR| LPC | LPC | ACL0 |LPC|| | | | | | ||0 0 0 0 0 0|0 1|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2||5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| ACL2 |ACL|A| GAIN0 |ACL|ACL| GAIN0 | GAIN1 || | 1 |C| | 3 | 2 | | ||0 0 0 0 0|0 0|0|0 0 0 0|0 0|0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0||4 3 2 1 0|1 0|6|3 2 1 0|1 0|6 5|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| GAIN2 | GAIN1 | GAIN2 | GAIN3 | GRID | GAIN3 || | | | | | ||0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0||3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|4 3 2 1|1 0 9 8|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| POS0 | POS1 | POS0 | POS1 | POS2 || | | | | ||0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0||7 6 5 4 3 2 1 0|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| POS3 | POS2 | POS3 | PSIG1 | PSIG0 | PSIG3 | PSIG2 || | | | | | | ||0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0||3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|3 2 1 0|3 2 1 0|3 2 1 0|3 2 1 0|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Figure 2: G.723 (5.3 kb/s) bit packing

The packing of G.723.1 SID (silence) frames, which are indicated by the header (HDR) bits havingthe pattern “1 0”, is depicted in Fig. 3.

0 1 2 30 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| LPC |HDR| LPC | LPC | GAIN |LPC|| | | | | | ||0 0 0 0 0 0|1 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2||5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Figure 3: G.723 SID mode bit packing



4.5.4 G726-40, G726-32, G726-24, and G726-16

ITU-T Recommendation G.726 describes, among others, the algorithm recommended for conversionof a single 64 kbit/s A-law or mu-law PCM channel encoded at 8,000 samples/sec to and from a40, 32, 24, or 16 kbit/s channel. The conversion is applied to the PCM stream using an AdaptiveDifferential Pulse Code Modulation (ADPCM) transcoding technique. The ADPCM representationconsists of a series of codewords with a one-to-one correspondence to the samples in the PCMstream. The G726 data rates of 40, 32, 24, and 16 kbit/s have codewords of 5, 4, 3, and 2 bits,respectively.

The 16 and 24 kbit/s encodings do not provide toll quality speech. They are designed for usedin overloaded Digital Circuit Multiplication Equipment (DCME). ITU-T G.726 recommends thatthe 16 and 24 kbit/s encodings should be alternated with higher data rate encodings to provide anaverage sample size of between 3.5 and 3.7 bits per sample.

The encodings of G.726 are here denoted as G726-40, G726-32, G726-24, and G726-16. Prior to1990, G721 described the 32 kbit/s ADPCM encoding, and G723 described the 40, 32, and 16kbit/s encodings. Thus, G726-32 designates the same algorithm as G721 in RFC 1890.

A stream of G726 codewords contains no information on the encoding being used, therefore tran-sitions between G726 encoding types are not permitted within a sequence of packed codewords.Applications must determine the encoding type of packed codewords from the RTP payload iden-tifier.

No payload-specific header information shall be included as part of the audio data. A stream ofG726 codewords must be packed into octets as follows: the first codeword is placed into the firstoctet such that the least significant bit of the codeword aligns with the least significant bit in theoctet, the second codeword is then packed so that its least significant bit coincides with the leastsignificant unoccupied bit in the octet. When a complete codeword cannot be placed into an octet,the bits overlapping the octet boundary are placed into the least significant bits of the next octet.Packing must end with a completely packed final octet. The number of codewords packed willtherefore be a multiple of 8, 2, 8, and 4 for G726-40, G726-32, G726-24, and G726-16, respectively.An example of the packing scheme for G726-32 codewords is as shown, where bit 7 is the leastsignificant bit of the first octet, and bit A3 is the least significant bit of the first codeword:

0 10 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-|B B B B|A A A A|D D D D|C C C C| ...|0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-

An example of the packing scheme for G726-24 codewords follows, where again bit 7 is the leastsignificant bit of the first octet, and bit A2 is the least significant bit of the first codeword:



0 1 20 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-|C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...|1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-

Note that the “little-endian” direction in which samples are packed into octets in the G726-16, -24,-32 and -40 payload formats specified here is consistent with ITU-T Recommendation X.420, butis the opposite of what is specified in ITU-T Recommendation I.366.2 Annex E for ATM AAL2transport. A second set of RTP payload formats matching the packetization of I.366.2 Annex Eand identified by MIME subtypes AAL2-G726-16, -24, -32 and -40 will be specified in a separatedocument.

4.5.5 G728

G728 is specified in ITU-T Recommendation G.728, “Coding of speech at 16 kbit/s using low-delaycode excited linear prediction”.

A G.278 encoder translates 5 consecutive audio samples into a 10-bit codebook index, resulting ina bit rate of 16 kb/s for audio sampled at 8,000 samples per second. The group of five consecutivesamples is called a vector. Four consecutive vectors, labeled V1 to V4 (where V1 is to be playedfirst by the receiver), build one G.728 frame. The four vectors of 40 bits are packed into 5 octets,labeled B1 through B5. B1 shall be placed first in the RTP packet.

Referring to the figure below, the principle for bit order is “maintenance of bit significance”. Bitsfrom an older vector are more significant than bits from newer vectors. The MSB of the frame goesto the MSB of B1 and the LSB of the frame goes to LSB of B5.

1 2 3 30 0 0 0 9++++++++++++++++++++++++++++++++++++++++<---V1---><---V2---><---V3---><---V4---> vectors<--B1--><--B2--><--B3--><--B4--><--B5--> octets<------------- frame 1 ---------------->

In particular, B1 contains the eight most significant bits of V1, with the MSB of V1 being the MSBof B1. B2 contains the two least significant bits of V1, the more significant of the two in its MSB,and the six most significant bits of V2. B1 shall be placed first in the RTP packet and B5 last.

4.5.6 G729

G729 is specified in ITU-T Recommendation G.729, “Coding of speech at 8 kbit/s using conjugatestructure-algebraic code excited linear prediction (CS-ACELP)”. A reduced-complexity version ofthe G.729 algorithm is specified in Annex A to Rec. G.729. The speech coding algorithms in themain body of G.729 and in G.729 Annex A are fully interoperable with each other, so there is no



need to further distinguish between them. An implementation that signals or accepts use of G729payload format may implement either G.729 or G.729A unless restricted by additional signalingspecified elsewhere related specifically to the encoding rather than the payload format. The G.729and G.729 Annex A codecs were optimized to represent speech with high quality, where G.729Annex A trades some speech quality for an approximate 50% complexity reduction [10]. See thenext Section (4.5.7) for other data rates added in later G.729 Annexes. For all data rates, thesampling frequency (and RTP timestamp clock rate) is 8,000 Hz.

A voice activity detector (VAD) and comfort noise generator (CNG) algorithm in Annex B ofG.729 is recommended for digital simultaneous voice and data applications and can be used inconjunction with G.729 or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets,while the G.729 Annex B comfort noise frame occupies 2 octets. Receivers must accept comfortnoise frames if restriction of their use has not been signaled. The MIME registration for G729 inRFC 3555 [7] specifies a parameter that may be used with MIME or SDP to restrict the use ofcomfort noise frames.

A G729 RTP packet may consist of zero or more G.729 or G.729 Annex A frames, followed by zeroor one G.729 Annex B frames. The presence of a comfort noise frame can be deduced from thelength of the RTP payload. The default packetization interval is 20 ms (two frames), but in somesituations it may be desirable to send 10 ms packets. An example would be a transition from speechto comfort noise in the first 10 ms of the packet. For some applications, a longer packetizationinterval may be required to reduce the packet rate.

0 1 2 30 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|L| L1 | L2 | L3 | P1 |P| C1 ||0| | | | |0| || |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2 3 4|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| C1 | S1 | GA1 | GB1 | P2 | C2 || 1 1 1| | | | | ||5 6 7 8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| C2 | S2 | GA2 | GB2 || 1 1 1| | | ||8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Figure 4: G.729 and G.729A bit packing

The transmitted parameters of a G.729/G.729A 10-ms frame, consisting of 80 bits, are defined inRecommendation G.729, Table 8/G.729. The mapping of the these parameters is given below inFig. 4. The diagrams show the bit packing in “network byte order”, also known as big-endianorder. The bits of each 32-bit word are numbered 0 to 31, with the most significant bit on theleft and numbered 0. The octets (bytes) of each word are transmitted most significant octet first.



The bits of each data field are numbered in the order as produced by the G.729 C code referenceimplementation.

The packing of the G.729 Annex B comfort noise frame is shown in Fig. 5.

0 10 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|L| LSF1 | LSF2 | GAIN |R||S| | | |E||F| | | |S||0|0 1 2 3 4|0 1 2 3|0 1 2 3 4|V| RESV = Reserved (zero)+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Figure 5: G.729 Annex B bit packing

4.5.7 G729D and G729E

Annexes D and E to ITU-T Recommendation G.729 provide additional data rates. Because thedata rate is not signaled in the bitstream, the different data rates are given distinct RTP encodingnames which are mapped to distinct payload type numbers. G729D indicates a 6.4 kbit/s codingmode (G.729 Annex D, for momentary reduction in channel capacity), while G729E indicates an11.8 kbit/s mode (G.729 Annex E, for improved performance with a wide range of narrow-bandinput signals, e.g., music and background noise). Annex E has two operating modes, backwardadaptive and forward adaptive, which are signaled by the first two bits in each frame (the mostsignificant two bits of the first octet).

The voice activity detector (VAD) and comfort noise generator (CNG) algorithm specified in AnnexB of G.729 may be used with Annex D and Annex E frames in addition to G.729 and G.729 AnnexA frames. The algorithm details for the operation of Annexes D and E with the Annex B CNGare specified in G.729 Annexes F and G. Note that Annexes F and G do not introduce any newencodings. Receivers must accept comfort noise frames if restriction of their use has not beensignaled. The MIME registrations for G729D and G729E in RFC 3555 [7] specify a parameter thatmay be used with MIME or SDP to restrict the use of comfort noise frames.

For G729D, an RTP packet may consist of zero or more G.729 Annex D frames, followed by zeroor one G.729 Annex B frame. Similarly, for G729E, an RTP packet may consist of zero or moreG.729 Annex E frames, followed by zero or one G.729 Annex B frame. The presence of a comfortnoise frame can be deduced from the length of the RTP payload.

A single RTP packet must contain frames of only one data rate, optionally followed by one comfortnoise frame. The data rate may be changed from packet to packet by changing the payload typenumber. G.729 Annexes D, E and H describe what the encoding and decoding algorithms must doto accommodate a change in data rate.

For G729D, the bits of a G.729 Annex D frame are formatted as shown below in Fig. 6 (cf. TableD.1/G.729). The frame length is 64 bits.



0 1 2 30 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|L| L1 | L2 | L3 | P1 | C1 ||0| | | | | || |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7|0 1 2 3 4 5|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| C1 |S1 | GA1 | GB1 | P2 | C2 |S2 | GA2 | GB2 || | | | | | | | | ||6 7 8|0 1|0 1 2|0 1 2|0 1 2 3|0 1 2 3 4 5 6 7 8|0 1|0 1 2|0 1 2|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Figure 6: G.729 Annex D bit packing

The net bit rate for the G.729 Annex E algorithm is 11.8 kbit/s and a total of 118 bits are used.Two bits are appended as “don’t care” bits to complete an integer number of octets for the frame.For G729E, the bits of a data frame are formatted as shown in the next two diagrams (cf. TableE.1/G.729). The fields for the G729E forward adaptive mode are packed as shown in Fig. 7.

0 1 2 30 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|0 0|L| L1 | L2 | L3 | P1 |P| C0_1|| |0| | | | |0| || | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| | C1_1 | C2_1 | C3_1 | C4_1 || | | | | ||3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| GA1 | GB1 | P2 | C0_2 | C1_2 | C2_2 || | | | | | ||0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| | C3_2 | C4_2 | GA2 | GB2 |DC || | | | | | ||6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Figure 7: G.729 Annex E (forward adaptive mode) bit packing

The fields for the G729E backward adaptive mode are packed as shown in Fig. 8.



0 1 2 30 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|1 1| P1 |P| C0_1 | C1_1 || | |0| 1 1 1| || |0 1 2 3 4 5 6 7|0|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| | C2_1 | C3_1 | C4_1 |GA1 | GB1 |P2 || | | | | | | ||8 9|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| | C0_2 | C1_2 | C2_2 || | 1 1 1| | ||2 3 4|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7 8 9|0 1 2 3 4 5|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+| | C3_2 | C4_2 | GA2 | GB2 |DC || | | | | | ||6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Figure 8: G.729 Annex E (backward adaptive mode) bit packing

4.5.8 GSM

GSM (Group Speciale Mobile) denotes the European GSM 06.10 standard for full-rate speechtranscoding, ETS 300 961, which is based on RPE/LTP (residual pulse excitation/long term pre-diction) coding at a rate of 13 kb/s [11, 12, 13]. The text of the standard can be obtained from:

ETSI (European Telecommunications Standards Institute)ETSI Secretariat: B.P.152F-06561 Valbonne CedexFrancePhone: +33 92 94 42 00Fax: +33 93 65 47 16

Blocks of 160 audio samples are compressed into 33 octets, for an effective data rate of 13,200 b/s.

4.5.8.1 General Packaging Issues

The GSM standard (ETS 300 961) specifies the bit stream produced by the codec, but does notspecify how these bits should be packed for transmission. The packetization specified here hassubsequently been adopted in ETSI Technical Specification TS 101 318. Some software implemen-tations of the GSM codec use a different packing than that specified here.



field field name bits field field name bits1 LARc[0] 6 39 xmc[22] 32 LARc[1] 6 40 xmc[23] 33 LARc[2] 5 41 xmc[24] 34 LARc[3] 5 42 xmc[25] 35 LARc[4] 4 43 Nc[2] 76 LARc[5] 4 44 bc[2] 27 LARc[6] 3 45 Mc[2] 28 LARc[7] 3 46 xmaxc[2] 69 Nc[0] 7 47 xmc[26] 310 bc[0] 2 48 xmc[27] 311 Mc[0] 2 49 xmc[28] 312 xmaxc[0] 6 50 xmc[29] 313 xmc[0] 3 51 xmc[30] 314 xmc[1] 3 52 xmc[31] 315 xmc[2] 3 53 xmc[32] 316 xmc[3] 3 54 xmc[33] 317 xmc[4] 3 55 xmc[34] 318 xmc[5] 3 56 xmc[35] 319 xmc[6] 3 57 xmc[36] 320 xmc[7] 3 58 xmc[37] 321 xmc[8] 3 59 xmc[38] 322 xmc[9] 3 60 Nc[3] 723 xmc[10] 3 61 bc[3] 224 xmc[11] 3 62 Mc[3] 225 xmc[12] 3 63 xmaxc[3] 626 Nc[1] 7 64 xmc[39] 327 bc[1] 2 65 xmc[40] 328 Mc[1] 2 66 xmc[41] 329 xmaxc[1] 6 67 xmc[42] 330 xmc[13] 3 68 xmc[43] 331 xmc[14] 3 69 xmc[44] 332 xmc[15] 3 70 xmc[45] 333 xmc[16] 3 71 xmc[46] 334 xmc[17] 3 72 xmc[47] 335 xmc[18] 3 73 xmc[48] 336 xmc[19] 3 74 xmc[49] 337 xmc[20] 3 75 xmc[50] 338 xmc[21] 3 76 xmc[51] 3

Table 2: Ordering of GSM variables



Octet Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 70 1 1 0 1 LARc0.0 LARc0.1 LARc0.2 LARc0.31 LARc0.4 LARc0.5 LARc1.0 LARc1.1 LARc1.2 LARc1.3 LARc1.4 LARc1.52 LARc2.0 LARc2.1 LARc2.2 LARc2.3 LARc2.4 LARc3.0 LARc3.1 LARc3.23 LARc3.3 LARc3.4 LARc4.0 LARc4.1 LARc4.2 LARc4.3 LARc5.0 LARc5.14 LARc5.2 LARc5.3 LARc6.0 LARc6.1 LARc6.2 LARc7.0 LARc7.1 LARc7.25 Nc0.0 Nc0.1 Nc0.2 Nc0.3 Nc0.4 Nc0.5 Nc0.6 bc0.06 bc0.1 Mc0.0 Mc0.1 xmaxc00 xmaxc01 xmaxc02 xmaxc03 xmaxc047 xmaxc05 xmc0.0 xmc0.1 xmc0.2 xmc1.0 xmc1.1 xmc1.2 xmc2.08 xmc2.1 xmc2.2 xmc3.0 xmc3.1 xmc3.2 xmc4.0 xmc4.1 xmc4.29 xmc5.0 xmc5.1 xmc5.2 xmc6.0 xmc6.1 xmc6.2 xmc7.0 xmc7.1

10 xmc7.2 xmc8.0 xmc8.1 xmc8.2 xmc9.0 xmc9.1 xmc9.2 xmc10.011 xmc10.1 xmc10.2 xmc11.0 xmc11.1 xmc11.2 xmc12.0 xmc12.1 xcm12.212 Nc1.0 Nc1.1 Nc1.2 Nc1.3 Nc1.4 Nc1.5 Nc1.6 bc1.013 bc1.1 Mc1.0 Mc1.1 xmaxc10 xmaxc11 xmaxc12 xmaxc13 xmaxc1414 xmax15 xmc13.0 xmc13.1 xmc13.2 xmc14.0 xmc14.1 xmc14.2 xmc15.015 xmc15.1 xmc15.2 xmc16.0 xmc16.1 xmc16.2 xmc17.0 xmc17.1 xmc17.216 xmc18.0 xmc18.1 xmc18.2 xmc19.0 xmc19.1 xmc19.2 xmc20.0 xmc20.117 xmc20.2 xmc21.0 xmc21.1 xmc21.2 xmc22.0 xmc22.1 xmc22.2 xmc23.018 xmc23.1 xmc23.2 xmc24.0 xmc24.1 xmc24.2 xmc25.0 xmc25.1 xmc25.219 Nc2.0 Nc2.1 Nc2.2 Nc2.3 Nc2.4 Nc2.5 Nc2.6 bc2.020 bc2.1 Mc2.0 Mc2.1 xmaxc20 xmaxc21 xmaxc22 xmaxc23 xmaxc2421 xmaxc25 xmc26.0 xmc26.1 xmc26.2 xmc27.0 xmc27.1 xmc27.2 xmc28.022 xmc28.1 xmc28.2 xmc29.0 xmc29.1 xmc29.2 xmc30.0 xmc30.1 xmc30.223 xmc31.0 xmc31.1 xmc31.2 xmc32.0 xmc32.1 xmc32.2 xmc33.0 xmc33.124 xmc33.2 xmc34.0 xmc34.1 xmc34.2 xmc35.0 xmc35.1 xmc35.2 xmc36.025 Xmc36.1 xmc36.2 xmc37.0 xmc37.1 xmc37.2 xmc38.0 xmc38.1 xmc38.226 Nc3.0 Nc3.1 Nc3.2 Nc3.3 Nc3.4 Nc3.5 Nc3.6 bc3.027 bc3.1 Mc3.0 Mc3.1 xmaxc30 xmaxc31 xmaxc32 xmaxc33 xmaxc3428 xmaxc35 xmc39.0 xmc39.1 xmc39.2 xmc40.0 xmc40.1 xmc40.2 xmc41.029 xmc41.1 xmc41.2 xmc42.0 xmc42.1 xmc42.2 xmc43.0 xmc43.1 xmc43.230 xmc44.0 xmc44.1 xmc44.2 xmc45.0 xmc45.1 xmc45.2 xmc46.0 xmc46.131 xmc46.2 xmc47.0 xmc47.1 xmc47.2 xmc48.0 xmc48.1 xmc48.2 xmc49.032 xmc49.1 xmc49.2 xmc50.0 xmc50.1 xmc50.2 xmc51.0 xmc51.1 xmc51.2

Table 3: GSM payload format

In the GSM packing used by RTP, the bits shall be packed beginning from the most significantbit. Every 160 sample GSM frame is coded into one 33 octet (264 bit) buffer. Every such bufferbegins with a 4 bit signature (0xD), followed by the MSB encoding of the fields of the frame. Thefirst octet thus contains 1101 in the 4 most significant bits (0-3) and the 4 most significant bits ofF1 (0-3) in the 4 least significant bits (4-7). The second octet contains the 2 least significant bitsof F1 in bits 0-1, and F2 in bits 2-7, and so on. The order of the fields in the frame is described inTable 2.

4.5.8.2 GSM Variable Names and Numbers

In the RTP encoding we have the bit pattern described in Table 3, where F.i signifies the ith bitof the field F, bit 0 is the most significant bit, and the bits of every octet are numbered from 0 to



7 from most to least significant.

4.5.9 GSM-EFR

GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding, specified in ETS 300 726which is available from ETSI at the address given in Section 4.5.8. This codec has a frame lengthof 244 bits. For transmission in RTP, each codec frame is packed into a 31 octet (248 bit) bufferbeginning with a 4-bit signature 0xC in a manner similar to that specified here for the originalGSM 06.10 codec. The packing is specified in ETSI Technical Specification TS 101 318.

4.5.10 L8

L8 denotes linear audio data samples, using 8-bits of precision with an offset of 128, that is, themost negative signal is encoded as zero.

4.5.11 L16

L16 denotes uncompressed audio data samples, using 16-bit signed representation with 65,535equally divided steps between minimum and maximum signal level, ranging from -32,768 to 32,767.The value is represented in two’s complement notation and transmitted in network byte order (mostsignificant byte first).

The MIME registration for L16 in RFC 3555 [7] specifies parameters that MAY be used with MIMEor SDP to indicate that analog pre-emphasis was applied to the signal before quantization or toindicate that a multiple-channel audio stream follows a different channel ordering convention thanis specified in Section 4.1.

4.5.12 LPC

LPC designates an experimental linear predictive encoding contributed by Ron Frederick, whichis based on an implementation written by Ron Zuckerman posted to the Usenet group comp.dspon June 26, 1992. The codec generates 14 octets for every frame. The framesize is set to 20 ms,resulting in a bit rate of 5,600 b/s.

4.5.13 MPA

MPA denotes MPEG-1 or MPEG-2 audio encapsulated as elementary streams. The encodingis defined in ISO standards ISO/IEC 11172-3 and 13818-3. The encapsulation is specified inRFC 2250 [14].

The encoding may be at any of three levels of complexity, called Layer I, II and III. The selectedlayer as well as the sampling rate and channel count are indicated in the payload. The RTPtimestamp clock rate is always 90,000, independent of the sampling rate. MPEG-1 audio supportssampling rates of 32, 44.1, and 48 kHz (ISO/IEC 11172-3, section 1.1; “Scope”). MPEG-2 supports



sampling rates of 16, 22.05 and 24 kHz. The number of samples per frame is fixed, but the framesize will vary with the sampling rate and bit rate.

The MIME registration for MPA in RFC 3555 [7] specifies parameters that MAY be used withMIME or SDP to restrict the selection of layer, channel count, sampling rate, and bit rate.

4.5.14 PCMA and PCMU

PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio data is encoded as eightbits per sample, after logarithmic scaling. PCMU denotes mu-law scaling, PCMA A-law scaling.A detailed description is given by Jayant and Noll [15]. Each G.711 octet shall be octet-alignedin an RTP packet. The sign bit of each G.711 octet shall correspond to the most significant bitof the octet in the RTP packet (i.e., assuming the G.711 samples are handled as octets on the hostmachine, the sign bit shall be the most significant bit of the octet as defined by the host machineformat). The 56 kb/s and 48 kb/s modes of G.711 are not applicable to RTP, since PCMA andPCMU must always be transmitted as 8-bit samples.

See Section 4.1 regarding silence suppression.

4.5.15 QCELP

The Electronic Industries Association (EIA) & Telecommunications Industry Association (TIA)standard IS-733, “TR45: High Rate Speech Service Option for Wideband Spread Spectrum Com-munications Systems”, defines the QCELP audio compression algorithm for use in wireless CDMAapplications. The QCELP CODEC compresses each 20 milliseconds of 8,000 Hz, 16-bit sampledinput speech into one of four different size output frames: Rate 1 (266 bits), Rate 1/2 (124 bits),Rate 1/4 (54 bits) or Rate 1/8 (20 bits). For typical speech patterns, this results in an averageoutput of 6.8 kb/s for normal mode and 4.7 kb/s for reduced rate mode. The packetization of theQCELP audio codec is described in [16].

4.5.16 RED

The redundant audio payload format “RED” is specified by RFC 2198 [17]. It defines a means bywhich multiple redundant copies of an audio packet may be transmitted in a single RTP stream.Each packet in such a stream contains, in addition to the audio data for that packetization interval,a (more heavily compressed) copy of the data from a previous packetization interval. This allows anapproximation of the data from lost packets to be recovered upon decoding of a subsequent packet,giving much improved sound quality when compared with silence substitution for lost packets.

4.5.17 VDVI

VDVI is a variable-rate version of DVI4, yielding speech bit rates of between 10 and 25 kb/s.It is specified for single-channel operation only. Samples are packed into octets starting at themost-significant bit. The last octet is padded with 1 bits if the last sample does not fill the last



octet. This padding is distinct from the valid codewords. The receiver needs to detect the paddingbecause there is no explicit count of samples in the packet.

It uses the following encoding:

DVI4 codeword VDVI bit pattern0 001 0102 11003 111004 1111005 11111006 111111007 111111108 109 011

10 110111 1110112 11110113 111110114 1111110115 11111111

5. Video

The following sections describe the video encodings that are defined in this memo and give theirabbreviated names used for identification. These video encodings and their payload types are listedin Table 5.

All of these video encodings use an RTP timestamp frequency of 90,000 Hz, the same as the MPEGpresentation time stamp frequency. This frequency yields exact integer timestamp increments forthe typical 24 (HDTV), 25 (PAL), and 29.97 (NTSC) and 30 Hz (HDTV) frame rates and 50,59.94 and 60 Hz field rates. While 90 kHz is the recommended rate for future video encodingsused within this profile, other rates may be used. However, it is not sufficient to use the videoframe rate (typically between 15 and 30 Hz) because that does not provide adequate resolution fortypical synchronization requirements when calculating the RTP timestamp corresponding to theNTP timestamp in an RTCP SR packet. The timestamp resolution must also be sufficient for thejitter estimate contained in the receiver reports.

For most of these video encodings, the RTP timestamp encodes the sampling instant of the videoimage contained in the RTP data packet. If a video image occupies more than one packet, thetimestamp is the same on all of those packets. Packets from different video images are distinguishedby their different timestamps.

Most of these video encodings also specify that the marker bit of the RTP header should be set toone in the last packet of a video frame and otherwise set to zero. Thus, it is not necessary to waitfor a following packet with a different timestamp to detect that a new frame should be displayed.



5.1 CelB

The CELL-B encoding is a proprietary encoding proposed by Sun Microsystems. The byte streamformat is described in RFC 2029 [18].

5.2 JPEG

The encoding is specified in ISO Standards 10918-1 and 10918-2. The RTP payload format is asspecified in RFC 2435 [19].

5.3 H261

The encoding is specified in ITU-T Recommendation H.261, “Video codec for audiovisual servicesat p x 64 kbit/s”. The packetization and RTP-specific properties are described in RFC 2032 [20].

5.4 H263

The encoding is specified in the 1996 version of ITU-T Recommendation H.263, “Video codingfor low bit rate communication”. The packetization and RTP-specific properties are described inRFC 2190 [21]. The H263-1998 payload format is recommended over this one for use by newimplementations.

5.5 H263-1998

The encoding is specified in the 1998 version of ITU-T Recommendation H.263, “Video codingfor low bit rate communication”. The packetization and RTP-specific properties are described inRFC 2429 [22]. Because the 1998 version of H.263 is a superset of the 1996 syntax, this payloadformat can also be used with the 1996 version of H.263, and is recommended for this use by newimplementations. This payload format does not replace RFC 2190, which continues to be used byexisting implementations, and may be required for backward compatibility in new implementations.Implementations using the new features of the 1998 version of H.263 must use the payload formatdescribed in RFC 2429.

5.6 MPV

MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary streams as specifiedin ISO Standards ISO/IEC 11172 and 13818-2, respectively. The RTP payload format is as specifiedin RFC 2250 [14], Section 3.

The MIME registration for MPV in RFC 3555 [7] specifies a parameter that MAY be used withMIME or SDP to restrict the selection of the type of MPEG video.



5.7 MP2T

MP2T designates the use of MPEG-2 transport streams, for either audio or video. The RTPpayload format is described in RFC 2250 [14], Section 2.

5.8 nv

The encoding is implemented in the program ‘nv’, version 4, developed at Xerox PARC by RonFrederick. Further information is available from the author:

Ron FrederickBlue Coat Systems Inc.650 Almanor AvenueSunnyvale, CA 94085United States

EMail: [email protected]

6. Payload Type Definitions

Tables 4 and 5 define this profile’s static payload type values for the PT field of the RTP dataheader. In addition, payload type values in the range 96-127 may be defined dynamically througha conference control protocol, which is beyond the scope of this document. For example, a sessiondirectory could specify that for a given session, payload type 96 indicates PCMU encoding, 8,000Hz sampling rate, 2 channels. Entries in Tables 4 and 5 with payload type “dyn” have no staticpayload type assigned and are only used with a dynamic payload type. Payload type 2 was assignedto G721 in RFC 1890 and to its equivalent successor G726-32 in draft versions of this specification,but its use is now deprecated and that static payload type is marked reserved due to conflicting usefor the payload formats G726-32 and AAL2-G726-32 (see Section 4.5.4). Payload type 13 indicatesthe Comfort Noise (CN) payload format specified in RFC 3389 [9]. Payload type 19 is marked“reserved” because some draft versions of this specification assigned that number to an earlierversion of the comfort noise payload format. The payload type range 72-76 is marked “reserved”so that RTCP and RTP packets can be reliably distinguished (see Section “Summary of ProtocolConstants” of the RTP protocol specification).

The payload types currently defined in this profile are assigned to exactly one of three categoriesor media types: audio only, video only and those combining audio and video. The media typesare marked in Tables 4 and 5 as “A”, “V” and “AV”, respectively. Payload types of differentmedia types shall not be interleaved or multiplexed within a single RTP session, but multipleRTP sessions may be used in parallel to send multiple media types. An RTP source may changepayload types within the same media type during a session. See the section “Multiplexing RTPSessions” of RFC 3550 for additional explanation.

Session participants agree through mechanisms beyond the scope of this specification on the set ofpayload types allowed in a given session. This set may, for example, be defined by the capabilities



PT encoding media type clock rate channelsname (Hz)

0 PCMU A 8,000 11 reserved A2 reserved A3 GSM A 8,000 14 G723 A 8,000 15 DVI4 A 8,000 16 DVI4 A 16,000 17 LPC A 8,000 18 PCMA A 8,000 19 G722 A 8,000 110 L16 A 44,100 211 L16 A 44,100 112 QCELP A 8,000 113 CN A 8,000 114 MPA A 90,000 (see text)15 G728 A 8,000 116 DVI4 A 11,025 117 DVI4 A 22,050 118 G729 A 8,000 119 reserved A20 unassigned A21 unassigned A22 unassigned A23 unassigned Adyn G726-40 A 8,000 1dyn G726-32 A 8,000 1dyn G726-24 A 8,000 1dyn G726-16 A 8,000 1dyn G729D A 8,000 1dyn G729E A 8,000 1dyn GSM-EFR A 8,000 1dyn L8 A var. var.dyn RED A (see text)dyn VDVI A var. 1

Table 4: Payload types (PT) for audio encodings



PT encoding media type clock ratename (Hz)

24 unassigned V25 CelB V 90,00026 JPEG V 90,00027 unassigned V28 nv V 90,00029 unassigned V30 unassigned V31 H261 V 90,00032 MPV V 90,00033 MP2T AV 90,00034 H263 V 90,00035-71 unassigned ?72-76 reserved N/A N/A77-95 unassigned ?96-127 dynamic ?dyn H263-1998 V 90,000

Table 5: Payload types (PT) for video and combined encodings

of the applications used, negotiated by a conference control protocol or established by agreementbetween the human participants.

Audio applications operating under this profile should, at a minimum, be able to send and/orreceive payload types 0 (PCMU) and 5 (DVI4). This allows interoperability without format nego-tiation and ensures successful negotiation with a conference control protocol.

7. RTP over TCP and Similar Byte Stream Protocols

Under special circumstances, it may be necessary to carry RTP in protocols offering a byte streamabstraction, such as TCP, possibly multiplexed with other data. The application must define itsown method of delineating RTP and RTCP packets (RTSP [23] provides an example of such anencapsulation specification).

8. Port Assignment

As specified in the RTP protocol definition, RTP data should be carried on an even UDP portnumber and the corresponding RTCP packets should be carried on the next higher (odd) portnumber.

Applications operating under this profile may use any such UDP port pair. For example, the portpair may be allocated randomly by a session management program. A single fixed port number



pair cannot be required because multiple applications using this profile are likely to run on thesame host, and there are some operating systems that do not allow multiple processes to use thesame UDP port with different multicast addresses.

However, port numbers 5004 and 5005 have been registered for use with this profile for thoseapplications that choose to use them as the default pair. Applications that operate under multipleprofiles may use this port pair as an indication to select this profile if they are not subject to theconstraint of the previous paragraph. Applications need not have a default and may require thatthe port pair be explicitly specified. The particular port numbers were chosen to lie in the rangeabove 5000 to accommodate port number allocation practice within some versions of the Unixoperating system, where port numbers below 1024 can only be used by privileged processes andport numbers between 1024 and 5000 are automatically assigned by the operating system.

9. Changes from RFC 1890

This RFC revises RFC 1890. It is mostly backwards-compatible with RFC 1890 except for functionsremoved because two interoperable implementations were not found. The additions to RFC 1890codify existing practice in the use of payload formats under this profile. Since this profile may beused without using any of the payload formats listed here, the addition of new payload formatsin this revision does not affect backwards compatibility. The changes are listed below, categorizedinto functional and non-functional changes.

Functional changes:

• Section 11, “IANA Considerations” was added to specify the registration of the name for thisprofile. That appendix also references a new Section 3 “Registering Additional Encodings”which establishes a policy that no additional registration of static payload types for this profilewill be made beyond those added in this revision and included in Tables 4 and 5. Instead,additional encoding names may be registered as MIME subtypes for binding to dynamicpayload types. Non-normative references were added to RFC 3555 [7] where MIME subtypesfor all the listed payload formats are registered, some with optional parameters for use of thepayload formats.

• Static payload types 4, 16, 17 and 34 were added to incorporate IANA registrations madesince the publication of RFC 1890, along with the corresponding payload format descriptionsfor G723 and H263.

• Following working group discussion, static payload types 12 and 18 were added along withthe corresponding payload format descriptions for QCELP and G729. Static payload type13 was assigned to the Comfort Noise (CN) payload format defined in RFC 3389. Payloadtype 19 was marked reserved because it had been temporarily allocated to an earlier versionof Comfort Noise present in some draft revisions of this document.

• The payload format for G721 was renamed to G726-32 following the ITU-T renumbering, andthe payload format description for G726 was expanded to include the -16, -24 and -40 datarates. Because of confusion regarding draft revisions of this document, some implementationsof these G726 payload formats packed samples into octets starting with the most significant bit



rather than the least significant bit as specified here. To partially resolve this incompatibility,new payload formats named AAL2-G726-16, -24, -32 and -40 will be specified in a separatedocument (see note in Section 4.5.4), and use of static payload type 2 is deprecated asexplained in Section 6.

• Payload formats G729D and G729E were added following the ITU-T addition of Annexes Dand E to Recommendation G.729. Listings were added for payload formats GSM-EFR, RED,and H263-1998 published in other documents subsequent to RFC 1890. These additionalpayload formats are referenced only by dynamic payload type numbers.

• The descriptions of the payload formats for G722, G728, GSM, VDVI were expanded.

• The payload format for 1016 audio was removed and its static payload type assignment 1 wasmarked “reserved” because two interoperable implementations were not found.

• Requirements for congestion control were added in Section 2.

• This profile follows the suggestion in the revised RTP spec that RTCP bandwidth may bespecified separately from the session bandwidth and separately for active senders and passivereceivers.

• The mapping of a user pass-phrase string into an encryption key was deleted from Section 2because two interoperable implementations were not found.

• The “quadrophonic” sample ordering convention for four-channel audio was removed to elim-inate an ambiguity as noted in Section 4.1.

Non-functional changes:

• In Section 4.1, it is now explicitly stated that silence suppression is allowed for all audiopayload formats. (This has always been the case and derives from a fundamental aspect ofRTP’s design and the motivations for packet audio, but was not explicit stated before.) Theuse of comfort noise is also explained.

• In Section 4.1, the requirement level for setting of the marker bit on the first packet aftersilence for audio was changed from “is” to “should be”, and clarified that the marker bit isset only when packets are intentionally not sent.

• Similarly, text was added to specify that the marker bit should be set to one on the lastpacket of a video frame, and that video frames are distinguished by their timestamps.

• RFC references are added for payload formats published after RFC 1890.

• The security considerations and full copyright sections were added.

• According to Peter Hoddie of Apple, only pre-1994 Macintosh used the 22254.54 rate andnone the 11127.27 rate, so the latter was dropped from the discussion of suggested samplingfrequencies.



• Table 1 was corrected to move some values from the “ms/packet” column to the “defaultms/packet” column where they belonged.

• Since the Interactive Multimedia Association ceased operations, an alternate resource wasprovided for a referenced IMA document.

• A note has been added for G722 to clarify a discrepancy between the actual sampling rateand the RTP timestamp clock rate.

• Small clarifications of the text have been made in several places, some in response to questionsfrom readers. In particular:

– A definition for “media type” is given in Section 1.1 to allow the explanation of multi-plexing RTP sessions in Section 6 to be more clear regarding the multiplexing of multiplemedia.

– The explanation of how to determine the number of audio frames in a packet from thelength was expanded.

– More description of the allocation of bandwidth to SDES items is given.

– A note was added that the convention for the order of channels specified in Section 4.1may be overridden by a particular encoding or payload format specification.

– The terms must, should, may, etc. are used as defined in RFC 2119.

• A second author for this document was added.

10. Security Considerations

Implementations using the profile defined in this specification are subject to the security consider-ations discussed in the RTP specification [1]. This profile does not specify any different securityservices. The primary function of this profile is to list a set of data compression encodings for audioand video media.

Confidentiality of the media streams is achieved by encryption. Because the data compressionused with the payload formats described in this profile is applied end-to-end, encryption may beperformed after compression so there is no conflict between the two operations.

A potential denial-of-service threat exists for data encodings using compression techniques thathave non-uniform receiver-end computational load. The attacker can inject pathological datagramsinto the stream which are complex to decode and cause the receiver to be overloaded.

As with any IP-based protocol, in some circumstances a receiver may be overloaded simply by thereceipt of too many packets, either desired or undesired. Network-layer authentication may be usedto discard packets from undesired sources, but the processing cost of the authentication itself maybe too high. In a multicast environment, source pruning is implemented in IGMPv3 (RFC 3376)[24] and in multicast routing protocols to allow a receiver to select which sources are allowed toreach it.



11. IANA Considerations

The RTP specification establishes a registry of profile names for use by higher-level control protocols,such as the Session Description Protocol (SDP), RFC 2327 [6], to refer to transport methods. Thisprofile registers the name “RTP/AVP”.

Section 3 establishes the policy that no additional registration of static RTP payload types for thisprofile will be made beyond those added in this document revision and included in Tables 4 and5. IANA may reference that section in declining to accept any additional registration requests. InTables 4 and 5, note that types 1 and 2 have been marked reserved and the set of “dyn” payloadtypes included has been updated. These changes are explained in Sections 6 and 9.

12. References

12.1 Normative References

[1] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, “RTP: A Transport Protocol forReal-Time Applications”, RFC 3550, July 2003.

[2] Bradner, S., “Key Words for Use in RFCs to Indicate Requirement Levels”, BCP 14, RFC 2119,March 1997.

[3] Apple Computer, “Audio Interchange File Format AIFF-C”, August 1991. (alsoftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).

12.2 Informative References

[4] Braden, R., Clark, D. and S. Shenker, “Integrated Services in the Internet Architecture: anOverview”, RFC 1633, June 1994.

[5] Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z. and W. Weiss, “An Architecture forDifferentiated Service”, RFC 2475, December 1998.

[6] Handley, M. and V. Jacobson, “SDP: Session Description Protocol”, RFC 2327, April 1998.

[7] Casner, S. and P. Hoschka, “MIME Type Registration of RTP Payload Types”, RFC 3555,July 2003.

[8] Freed, N., Klensin, J. and J. Postel, “Multipurpose Internet Mail Extensions (MIME) PartFour: Registration Procedures”, BCP 13, RFC 2048, November 1996.

[9] Zopf, R., “Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)”, RFC 3389,September 2002.

[10] Deleam, D. and J.-P. Petit, “Real-time implementations of the recent ITU-T low bit ratespeech coders on the TI TMS320C54X DSP: results, methodology, and applications”, in Proc.of International Conference on Signal Processing, Technology, and Applications (ICSPAT),(Boston, Massachusetts), pp. 1656–1660, October 1996.



[11] Mouly, M. and M.-B. Pautet, The GSM system for mobile communications. Lassay-les-Chateaux, France: Europe Media Duplication, 1993.

[12] Degener, J., “Digital Speech Compression”, Dr. Dobb’s Journal, December 1994.

[13] Redl, S., Weber, M. and M. Oliphant, An Introduction to GSM. Boston: Artech House, 1995.

[14] Hoffman, D., Fernando, G., Goyal, V. and M. Civanlar, “RTP Payload Format forMPEG1/MPEG2 Video”, RFC 2250, January 1998.

[15] Jayant, N. and P. Noll, Digital Coding of Waveforms—Principles and Applications to Speechand Video. Englewood Cliffs, New Jersey: Prentice-Hall, 1984.

[16] McKay, K., “RTP Payload Format for PureVoice(tm) Audio”, RFC 2658, August 1999.

[17] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M., Bolot, J.-C., Vega-Garcia,A. and S. Fosse-Parisis, “RTP Payload for Redundant Audio Data”, RFC 2198, September1997.

[18] Speer, M. and D. Hoffman, “RTP Payload Format of Sun’s CellB Video Encoding”, RFC 2029,October 1996.

[19] Berc, L., Fenner, W., Frederick, R., McCanne, S. and P. Stewart, “RTP Payload Format forJPEG-Compressed Video”, RFC 2435, October 1998.

[20] Turletti, T. and C. Huitema, “RTP Payload Format for H.261 Video Streams”, RFC 2032,October 1996.

[21] Zhu, C., “RTP Payload Format for H.263 Video Streams”, RFC 2190, September 1997.

[22] Bormann, C., Cline, L., Deisher, G., Gardos, T., Maciocco, C., Newell, D., Ott, J., Sullivan,G., Wenger, S. and C. Zhu, “RTP Payload Format for the 1998 Version of ITU-T Rec. H.263Video (H.263+)”, RFC 2429, October 1998.

[23] Schulzrinne, H., Rao, A. and R. Lanphier, “Real Time Streaming Protocol (RTSP)”, RFC 2326,April 1998.

[24] Cain, B., Deering, S., Kouvelas, I., Fenner, B. and A. Thyagarajan, “Internet Group Manage-ment Protocol, Version 3”, RFC 3376, October 2002.

13. Current Locations of Related Resources

Note: Several sections below refer to the ITU-T Software Tool Library (STL). It is availablefrom the ITU Sales Service, Place des Nations, CH-1211 Geneve 20, Switzerland (also checkhttp://www.itu.int). The ITU-T STL is covered by a license defined in ITU-T RecommendationG.191, “Software tools for speech and audio coding standardization”.



DVI4

An archived copy of the document IMA Recommended Practices for Enhancing Digital AudioCompatibility in Multimedia Systems (version 3.0), which describes the IMA ADPCM algorithm,is available at:

http://www.cs.columbia.edu/~hgs/audio/dvi/

An implementation is available from Jack Jansen at

ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar

G722

An implementation of the G.722 algorithm is available as part of the ITU-T STL, described above.

G723

The reference C code implementation defining the G.723.1 algorithm and its Annexes A, B, and Care available as an integral part of Recommendation G.723.1 from the ITU Sales Service, addresslisted above. Both the algorithm and C code are covered by a specific license. The ITU-T Secretariatshould be contacted to obtain such licensing information.

G726

G726 is specified in the ITU-T Recommendation G.726, “40, 32, 24, and 16 kb/s Adaptive Differ-ential Pulse Code Modulation (ADPCM)”. An implementation of the G.726 algorithm is availableas part of the ITU-T STL, described above.

G729

The reference C code implementation defining the G.729 algorithm and its Annexes A through I areavailable as an integral part of Recommendation G.729 from the ITU Sales Service, listed above.Annex I contains the integrated C source code for all G.729 operating modes. The G.729 algorithmand associated C code are covered by a specific license. The contact information for obtaining thelicense is available from the ITU-T Secretariat.

GSM

A reference implementation was written by Carsten Bormann and Jutta Degener (then at TUBerlin, Germany). It is available at

http://www.dmn.tzi.org/software/gsm/



Although the RPE-LTP algorithm is not an ITU-T standard, there is a C code implementationof the RPE-LTP algorithm available as part of the ITU-T STL. The STL implementation is anadaptation of the TU Berlin version.

LPC

An implementation is available at

ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z

PCMU, PCMA

An implementation of these algorithms is available as part of the ITU-T STL, described above.

14. Acknowledgments

The comments and careful review of Simao Campos, Richard Cox and AVT Working Group par-ticipants are gratefully acknowledged. The GSM description was adopted from the IMTC Voiceover IP Forum Service Interoperability Implementation Agreement (January 1997). Fred Burg andTerry Lyons helped with the G.729 description.

15. Intellectual Property Rights Statement

The IETF takes no position regarding the validity or scope of any intellectual property or otherrights that might be claimed to pertain to the implementation or use of the technology described inthis document or the extent to which any license under such rights might or might not be available;neither does it represent that it has made any effort to identify any such rights. Information on theIETF’s procedures with respect to rights in standards-track and standards-related documentationcan be found in BCP-11. Copies of claims of rights made available for publication and any assur-ances of licenses to be made available, or the result of an attempt made to obtain a general licenseor permission for the use of such proprietary rights by implementors or users of this specificationcan be obtained from the IETF Secretariat.

The IETF invites any interested party to bring to its attention any copyrights, patents or patentapplications, or other proprietary rights which may cover technology that may be required topractice this standard. Please address the information to the IETF Executive Director.



16. Authors’ Addresses

Henning SchulzrinneDepartment of Computer ScienceColumbia University1214 Amsterdam AvenueNew York, NY 10027United States


Stephen L. CasnerPacket Design3400 Hillview Avenue, Building 3Palo Alto, CA 94304United States




17. Full Copyright Statement

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This document and translations of it may be copied and furnished to others, and derivative worksthat comment on or otherwise explain it or assist in its implementation may be prepared, copied,published and distributed, in whole or in part, without restriction of any kind, provided that theabove copyright notice and this paragraph are included on all such copies and derivative works.However, this document itself may not be modified in any way, such as by removing the copyrightnotice or references to the Internet Society or other Internet organizations, except as needed forthe purpose of developing Internet standards in which case the procedures for copyrights definedin the Internet Standards process must be followed, or as required to translate it into languagesother than English.

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Acknowledgement

Funding for the RFC Editor function is currently provided by the Internet Society.


RTP Pro le for Audio and Video Conferences with Minimal Control

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